PortSIP Knowledge Base
  • PortSIP Communications Solution
    • What is PortSIP?
    • The Advantages of PortSIP PBX vs. Other PBXs
    • Cloud PBX Empowering Service Providers
    • Simplifying Unified Communications with WebRTC and SIP
    • What is CPaaS? Communications Platform as a Service Explained
    • UCaaS is Unified Communications as a Service
    • PortSIP PBX Administration Guide
      • Overview
      • PortSIP Security Features
      • Summary of Changes
      • Before Started
      • 1 Installation of the PortSIP PBX
        • Installation of PortSIP PBX v22.x
          • Install PortSIP PBX on Linux
          • Install PortSIP IM Server on Linux
          • Install PortSIP PBX on Windows
          • Upgrade to the Latest Version Within v22.x on Linux
          • Upgrade to the Latest v22.x on Windows
          • Upgrade v16.x to the Latest v22.x on Linux
        • Installation of PortSIP PBX v16.x
          • Upgrade to the Latest v16.x Release
      • 2 Configuring the PortSIP PBX
      • 3 Tenant Management
      • 4 Phone Device Management
        • Managing Phones
        • Auto Provisioning Security
        • Custom IP Phone Template
        • Bulk Importing Users and Auto Provisioning IP Phones
        • Zero Touch Provisioning Phones
        • Provision Phone Using PnP
        • PnP Auto Provisioning IP Phone Multicast Debug
        • Provision Phone Using RPS
        • Provision Phone Using DHCP Option 66
        • Provision Phone Using TFTP
        • Provisioning Cisco 79xx IP Phones
        • Provision Fanvil DECT IP Phones
        • Provision Yealink DECT IP Phones
        • Provision SNOM DECT IP Phones
        • Configuring Private RPS Account
      • 5 User Management
        • Users
        • How to Configure the Endpoints?
        • User Groups
        • DND and Automatic Callback
        • Speed Dial 8
        • Speed Dial 100
      • 6 Transport Management
      • 7 Trunk Management
        • Configuring SIP Trunk
        • Handle Outbound Calls Through SIP Trunk
      • 8 Call Route Management
        • Configuring Inbound Rule
        • Configuring Outbound Rule
      • 9 Configuring PortSIP SBC
        • Topology
        • Summary of Changes
        • Installation PortSIP SBC v11.x
        • Installation PortSIP SBC v10.x
        • Configuring PortSIP SBC for WebRTC
        • Upgrade to the Latest v11.x Release
        • Upgrade to the Latest v10.x Release
      • 10 Configuring SBC for MS Teams
        • Architecture
        • Configuring Microsoft Teams
        • Configuring SBC and PBX
        • Configure an SBC for Multiple Tenants
      • 11 Deploy the SBC Cluster
      • 12 Configuring Virtual Receptionist
        • Managing Virtual Receptionist
        • Visual IVR Editor Guide
        • Direct Inward System Access (DISA)
      • 13 Configuring Ring Group
      • 14 Call Parking
        • PortSIP Call Parking Feature
        • Using Call Parking Feature
        • Using Enhanced Call Park on Fanvil IP Phones
        • Using Enhanced Call Park on Yealink IP Phones
        • Using Enhanced Call Park on Grandstream IP Phones
        • Using Enhanced Call Park on Dinstar IP Phones
      • 15 Shared Voicemail
      • 16 Call Queue
        • Configuring Call Queue
        • Configuring Queue Callback
        • Agent States and Work Modes
        • Skills-Based Routing
        • Silent Monitoring
        • Wallboards
      • 17 Roles and Permissions
      • 18 E164 Number Processing
      • 19 Billing
      • 20 CDR and Call Recordings
        • CDR
        • Call Recordings
        • CDR Field Descriptions
      • 21 Call Reports
      • 22 Dealers
      • 23 Feature Access Codes
      • 24 Call Pickup
      • 25 Meetings
        • Joining a Meeting with the Invite Link
      • 26 Hot Desking
      • 27 STIR/SHAKEN
        • Configuring STIR/SHAKEN
      • 28 Digital Engagement Channels
        • SMS Channel
        • WhatsApp Channel
        • Manage SMS/WhatsApp Message Conversations
      • 29 Integrations
        • Microsoft 365 Integration
        • Google Workspace Integration
      • 30 Office Hours and Holiday Schedule
        • Configuring Office Hours and Holiday Schedule
        • Routing Calls Based on Office Hours and Holidays
      • 31 Configuring Email Notifications
      • PBX and SIP Trunk using PortSIP SBC
      • SIP Header Manipulation
      • Rebranding PortSIP PBX, SBC
      • System Service Extension Numbers
      • Certificates for TLS/HTTPS/WebRTC
        • Preparing TLS Certificates
        • Update Certificates
      • Backup and Restore: An Essential Guide
        • Backup and Restore PortSIP PBX
        • Backup and Restore PortSIP SBC
      • Storing Into AWS S3
      • Storing Into Azure Blob Storage
      • Trace Server - A Better Way to Monitoring SIP Messages and QoS for PortSIP PBX
    • Configuring SIP Trunks
      • QuestBlue SIP Trunk
        • Purchase a DID on QuestBlue Platform
        • Configuring QuestBlue IP Authentication Trunk
        • Configuring QuestBlue Register Authentication Trunk
        • Configuring Outbound & Inbound Calls
        • QuestBlue SMS Integration
      • Twilio SIP Trunk
        • Purchase a DID on the Twilio
        • Configuring Twilio Register Based Trunk
        • Configuring Twilio Interconnect Trunk
        • Configuring Outbound & Inbound Calls
        • Twilio SMS Integration
      • Telnyx SIP Trunk
        • Purchase a DID on Telnyx Platform
        • Configuring Telnyx IP Authentication Trunk
        • Configuring Telnyx Register Authentication Trunk
        • Configuring Outbound & Inbound Calls
        • Telnyx SMS Integration
      • Vonage SIP Trunk
        • Purchase a DID on Vonage Platform
        • Configuring Vonage IP Authentication Trunk
        • Configuring Vonage Register Authentication Trunk
        • Configuring Outbound & Inbound Calls
        • Vonage SMS Integration
      • VoIP.ms SIP Trunk
        • Purchase a DID on VoIP.ms
        • Configuring VoIP.ms Register Based Trunk
        • Configuring Outbound & Inbound Calls
        • VoIP.ms SMS Integration
      • Voxtelesys SIP Trunk
        • Purchase a DID on Voxtelesys Platform
        • Configuring Voxtelesys IP Authentication Trunk
        • Configuring Voxtelesys Register Authentication Trunk
        • Configuring Outbound & Inbound Calls
        • Voxtelesys SMS Integration
      • Wavix SIP Trunk
        • Purchase a DID on Wavix Platform
        • Configuring Wavix IP Authentication Trunk
        • Configuring Wavix Digest Trunk
        • Configuring Outbound & Inbound Calls
        • Wavix SMS Integration
      • VoIP Innovations SIP Trunk
        • Purchase a DID on VoIP Innovations Platform
        • Configuring VoIP Innovations IP Authentication Trunk
        • Configuring Outbound & Inbound Calls
        • VoIP Innovations SMS Integration
      • Bandwidth SIP Trunk
        • Purchase a DID on Bandwidth Platform
        • Configuring Bandwidth IP Authentication Trunk
        • Configuring Outbound & Inbound Calls
        • Bandwidth SMS Integration
      • Flowroute SIP Trunk
        • Purchase a DID on Flowroute Platform
        • Configuring Flowroute IP Authentication Trunk
        • Configuring Outbound & Inbound Calls
        • Flowroute SMS Integration
      • Gamma SIP Trunk
      • Aire Networks SIP Trunk
      • VoiceMeUp SIP Trunk
        • Configuring VoiceMeUp Trunk
        • VoiceMeUp SMS Integration
    • PBX Cluster (v22.x)
      • Topology
      • Preparing Cluster Servers
      • Configuring Cluster Servers
      • Managing Cluster
    • High Availability (v22.x)
      • High Availability and Sclability On-Premise
        • PortSIP PBX High Availability Architecture
        • High Availability Installations on Ubuntu
        • Upgrading High Availability Installation
        • Scaling Servers On-Premise for High Availability
        • Scaling SBC On-Premise for High Availability
        • Scaling IM Server On-Premise for High Availability
      • High Availability and Scalability on AWS
        • PortSIP PBX High Availability Architecture
        • High Availability Installations on AWS
        • Upgrading High Availability Installation
        • Increase Size of EBS Volume
        • Scaling Servers on AWS for High Availability
    • PBX Cluster(v16.x)
      • Topology
      • Preparing Cluster Servers
      • Configuring Cluster Servers
      • Managing Cluster
      • Configuring Cluster Servers for High Availability
    • High Availability (v16.x)
      • High Availability for On-Premise
        • PortSIP PBX High Availability Architecture
        • High Availability Installations on Ubuntu
        • Upgrading High Availability Installation
      • High Availability and Scalability on AWS
        • PortSIP PBX High Availability Architecture
        • High Availability Installations on AWS
        • Upgrading High Availability Installations
        • Increase Size of EBS Volume
        • Scaling Servers on AWS for HA
    • PortSIP UCaaS
    • FAQ
      • Troubleshooting Call Issues
      • How to Activate License key?
      • What is the Multi-Tenant PBX?
      • Is the PortSIP PBX built on Asterisk, FreeSwitch?
      • What is the SBC?
      • What is the PBX? Features, Benefits
      • What File Format Is Required for PortSIP PBX Prompt?
      • What is Direct Inward Dialing (DID)?
      • What is the DID Pool?
      • What are IP Phones Work with PortSIP PBX?
      • Hardware Specifications
      • How to Adjust the REST API Rate Limit?
      • SIP Status Code of Response
      • What is SIP ALG and Why You Need to Disable It?
      • Essential Factors for Choosing a Could PBX Solution
      • Migrate from legacy FCM APIs to HTTP v1 for Android Push Notifications
      • PortSIP SDK License Agreement
      • PortSIP Software End-User License Agreement
  • Apps Guides
    • PortSIP ONE Desktop App
      • Sign in to PBX
      • Calls, Messages, and Voicemails
      • Customize Your Caller ID
      • SMS and WhatsApp Messaging
      • Click to Call
      • Calling from Another Device
      • Change Your Call Queue Status
    • PortSIP ONE Mobile App
      • Sign in to PBX
      • Calls, Messages, and Voicemails
      • Customize Your Caller ID
      • SMS and WhatsApp Messaging
      • Change Your Call Queue Status
    • PortSIP Softphone
  • DEVELOPING WITH PORTSIP
    • Getting Started
    • Calling APIs
      • User Manual for Windows
      • User Manual for iOS
      • User Manual for Android
      • User Manual for macOS
    • REST APIs
      • Version 22.0
        • About
        • API reference
          • Info
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          • Sessions
            • Directly
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            • Participants
              • Layout
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              • Chairman
              • Position
              • Destroy
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              • Set read
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            • Export
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          • Files
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            • Export
          • Moh server
            • Musics
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          • Phone models
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            • Members
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            • Export
            • Assignees
              • Destroy
          • Ring groups
            • Agents
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          • Shared voicemails
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            • Voicemails
              • Set read
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            • Greetings
              • Enable
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              • Destroy
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            • Destroy
          • Allowed country codes
          • Disallowed codes
            • Destroy
            • Export
          • Blacklisted numbers
            • Destroy
            • Export
          • Call rates
            • Destroy
            • Export
          • Ivr servers
            • Status
            • Destroy
          • Ivrs
            • Status
            • Destroy
            • Action urls
              • Destroy
          • Hotdesking
            • Status
            • Logout
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          • Sms
            • Destroy
          • Whatsapp
            • Destroy
          • Cdrs
          • Call reports
            • Destroy
          • Completed call reports
            • Destroy
          • Feature access codes
          • Default email templates
          • Custom email templates
          • Audit logs
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          • Ms365
            • Certificate
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              • Destroy
        • Specification
      • Version 22.1
        • About
        • API reference
          • Info
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            • By microsoft
          • Logout
          • Network
          • Sbc
            • Token
              • Destroy
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            • Destroy
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              • Mute
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              • Destroy
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              • Favorite
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                • Diff
            • Call queues
              • Agent
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            • Business contacts
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            • Destroy
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          • Contacts
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            • Musics
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            • Managers
              • Destroy
          • Call park
          • Call park groups
            • Destroy
            • Members
              • Destroy
          • Call pickup groups
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            • Members
          • Voicemail
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          • Outbound rules
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            • Export
            • Applied groups
              • Destroy
          • Phone models
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            • Reprovision
            • Assignee
            • Reboot
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          • Dect phone models
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            • Members
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              • Destroy
          • Ring groups
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              • Set read
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          • Call rates
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            • Action urls
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          • Cdrs
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  • Best Practices
  • Template Columns Explanation
  • Sample CSV File
  1. PortSIP Communications Solution
  2. PortSIP PBX Administration Guide
  3. 4 Phone Device Management

Bulk Importing Users and Auto Provisioning IP Phones

PreviousCustom IP Phone TemplateNextZero Touch Provisioning Phones

Last updated 4 months ago

This guide is designed to assist company administrators in configuring a large number of company employees using the bulk importing feature of PortSIP PBX. This feature allows for the quick and easy addition of users in a spreadsheet format, eliminating the need for time-consuming individual device setting configurations. Furthermore, when importing users in bulk, you can also auto-provision IP Phones to the users simultaneously.

Best Practices

  1. Create a user with all necessary settings, such as forwarding rules, office hours, IP Phone, and BLF keys.

  2. Export this user to a CSV file to serve as a template.

  3. Delete that user in the PortSIP PBX web portal.

  4. Edit the template CSV file to add new user information. This can be done by copying from the example user and making necessary changes. For instance, you may need to change the username, password, extension number, office hours, and phone information, among other things.

  5. Save your changes and then import the CSV file into the PortSIP PBX web portal.

By following these steps, you can efficiently manage and configure multiple users on the PortSIP PBX platform. This process not only saves time but also ensures consistency in user settings across the board. Remember, the key to successful bulk importing and auto-provisioning lies in careful preparation and attention to detail when creating and editing your CSV file.

The CSV file configuration in a spreadsheet or text editor must be performed in UTF-8 format.

Template Columns Explanation

The CSV file header columns like the below:

  • name: This is the username that the user will enter to access the PBX web portal. It should consist of numbers and letters.

  • enabled: This indicates whether this user is enabled or not. The value can be TRUE or FALSE. The user will be disabled if it is set to FALSE.

  • password: This is the password for the user to log in to the PBX web portal. It must meet the tenant’s password policy, otherwise, the importing will fail.

  • extension_number: This is the extension number of the user. It only accepts numbers and is limited to a maximum of 64 digits.

  • extension_password: This is the password for the extension to register to PBX from the SIP endpoint. It must meet the tenant’s password policy, otherwise, the importing will fail.

  • email: This is the email address of the user.

  • display_name: This is the full name of the user, for example: Jim Keeny.

  • enable_audio_recording: This indicates whether to enable the audio call recording or not for the user. TRUE for enabled and FALSE for disabled.

  • enable_video_recording: This indicates whether to enable the video recording or not for the user. TRUE for enabled and FALSE for disabled.

  • enable_acb: This indicates whether to enable the automatic callback or not for the user. TRUE for enabled and FALSE for disabled.

  • enable_dnd: This indicates whether to enable the Do Not Disturb or not for the user. TRUE for enabled and FALSE for disabled.

  • enable_hot_desking: This indicates whether to enable the Hot Desking or not for the user. TRUE for enabled and FALSE for disabled.

  • office_hours: This specifies the office hours for the user. It can use the global office hours from the tenant level, or create custom office hours for this user only. The office hours are defined in JSON format.

If the mode is "CUSTOM", it means to use the specified office hours for that user. For each weekday, if the key enabled is true, it means that the day is open, and the ranges is a JSON array used to define the time shifts for office hours. If the ranges is empty, it means the whole day is opened; if the key enabled is false, it means that the whole day is closed, and the ranges will be ignored.

{
	"mode": "CUSTOM",
	"monday": {
		"enabled": true,
		"ranges": [{
			"from": "09:00",
			"to": "11:00"
		}, {
			"from": "12:00",
			"to": "17:00"
		}]
	},
	"tuesday": {
		"enabled": true,
		"ranges": [{
			"from": "09:00",
			"to": "17:00"
		}]
	},
	"wednesday": {
		"enabled": true,
		"ranges": []
	},
	"thursday": {
		"enabled": true,
		"ranges": []
	},
	"friday": {
		"enabled": true,
		"ranges": []
	},
	"saturday": {
		"enabled": false,
		"ranges": []
	},
	"sunday": {
		"enabled": false,
		"ranges": []
	}
}

If the mode is set to "GLOBAL", it means that the user's office hours will follow the tenant's office hours.

{
	"mode": "GLOBAL",
	"monday": {
		"enabled": true,
		"ranges": []
	},
	"tuesday": {
		"enabled": true,
		"ranges": []
	},
	"wednesday": {
		"enabled": true,
		"ranges": []
	},
	"thursday": {
		"enabled": true,
		"ranges": []
	},
	"friday": {
		"enabled": true,
		"ranges": []
	},
	"saturday": {
		"enabled": true,
		"ranges": []
	},
	"sunday": {
		"enabled": true,
		"ranges": []
	}
}
  • anonymous_outbound_calls: This indicates whether to enable the anonymous call or not for the user. TRUE for enabled and FALSE for disabled.

  • delivery_outbound_cid: This indicates whether to rewrite the user part of the From header by the outbound caller ID in the SIP INVITE message or not when the user makes a call to the trunk. TRUE for enabled and FALSE for disabled.

  • sms: This indicates the permission for sending SMS and WhatsApp messages, it can be DISABLE, ALLOW_WITH_SENDER_ID, ALLOW. It's only applicable to the PBX v22.0 and higher.

  • voicemail_prompt: This specifies the voicemail prompt language. It’s BCP 47 Language Tags (The Internet Best Current Practices (BCP) for language tags), for example, en-US.

  • enable_voicemail_pin: This indicates whether to verify the voicemail PIN or not when the user accesses his voicemail by dialing FAC. TRUE for enabled and FALSE for disabled.

  • voicemail_pin: This sets the voicemail PIN. This is a mandatory field and must be numeric and meet the voicemail password policy.

  • voicemail_play_datetime: This indicates whether to play the date and time or not when the user accesses his voicemail. TRUE for enabled and FALSE for disabled.

  • enable_voicemail_notify: This indicates whether to send an email notification to the user’s email or not if a new voicemail is generated. TRUE for enabled and FALSE for disabled.

  • online_no_answer_forward_rule: This specifies the forward rule if the user is online but does not answer the call at a certain time. It’s a JSON object that includes the following keys:

    • action: Allows to set the FORWARD_TO_NUMBER, FORWARD_TO_VOICEMAIL, HANGUP.

    • timeout: Specifies the max ringing time before taking the action, in seconds.

    • number: Specifies the destination number for the action is set to FORWARD_TO_NUMBER. It will be ignored if the action is not set as FORWARD_TO_NUMBER.

For example, if the call is not answered in 60 seconds, that call will be forwarded to the voicemail. This can be set as follows:

{
	"action": "FORWARD_TO_VOICEMAIL",
	"timeout": 60,
	"number": ""
}
  • online_busy_forward_rule: This specifies the forward rule if the user is online but currently on a call. It’s a JSON object that includes the following keys:

    • action: Allows to set the FORWARD_TO_NUMBER, FORWARD_TO_VOICEMAIL, RING_ANYWAY, HANGUP.

    • timeout: It will be ignored for the online busy forward rule.

    • number: Specifies the destination number for the action is set to FORWARD_TO_NUMBER. It will be ignored if the action is not set as FORWARD_TO_NUMBER.

For example, if the user is on a call, the new call that called him will be sent to him anyway. The timeout and number keys will be ignored. This can be set as follows:

{
	"action": "RING_ANYWAY",
	"timeout": 60,
	"number": ""
}
  • offline_office_hours_forward_rule: This specifies the forward rule if the user is offline and currently in office hours. It’s a JSON object that includes the following keys:

    • action: Allows to set the FORWARD_TO_NUMBER, FORWARD_TO_VOICEMAIL, HANGUP.

    • timeout: It will be ignored for the offline forward rule.

    • number: Specifies the destination number for the action is set to FORWARD_TO_NUMBER. It will be ignored if the action is not set as FORWARD_TO_NUMBER.

For example, if the user is offline during office hours, the new call that called him will be hung up. The timeout and number keys will be ignored. This can be set as follows:

{
	"action": "HANGUP",
	"timeout": 60,
	"number": ""
}
  • offline_non_office_hours_forward_rule: This specifies the forward rule if the user is offline and currently outside of office hours. It’s a JSON object that includes the following keys:

    • action: Allows to set the FORWARD_TO_NUMBER, FORWARD_TO_VOICEMAIL, HANGUP.

    • timeout: It will be ignored for the offline forward rule.

    • number: Specifies the destination number for the action is set to FORWARD_TO_NUMBER. It will be ignored if the action is not set as FORWARD_TO_NUMBER.

For example, if the user is offline outside of office hours, the new call that called him will be forwarded to the number 123456. The timeout key will be ignored. This can be set as follows:

{
	"action": "FORWARD_TO_NUMBER",
	"timeout": 60,
	"number": "123456"
}
  • custom_forward_rules: These rules serve as exceptions that override the standard forwarding rules. They are defined as a JSON object. If you wish to leave this field empty, set it to []. This allows the user to establish their own exception rules via the PortSIP PBX web portal.

  • blfs: This field is used to specify the BLF (Busy Lamp Field) keys. It is also a JSON object. To leave it empty, set it to [].

  • interface: This field is used to specify the network interface IP Address, which is used to generate the QR code. The client application will connect to the PortSIP PBX using this IP address. It accepts the following values:

    • WEB_DOMAIN: The client app will use the PortSIP PBX Web Domain as the outbound proxy server address.

    • PUBLIC_IPV4: The client app will use the PortSIP public IP IPv4 address as the outbound proxy server address.

    • PUBLIC_IPV6: The client app will use the PortSIP PBX public IPv6 address as the outbound proxy server address.

    • PRIVATE_IPV4: The client app will use the PortSIP PBX private IPv4 address as the outbound proxy server address.

    • PRIVATE_IPV6: The client app will use the PortSIP PBX private IPv6 address as the outbound proxy server address.

    • SBC_DOMAIN: The client app will use the PortSIP SBC web domain as the outbound proxy server address, and then the client app registers to PortSIP PBX via the PortSIP SBC.

  • preferred_transport: This field is used to specify the transport protocol for the PortSIP PBX in the QR code. After scanning the QR code, the client app will prioritize using this specified transport to register with PortSIP PBX. This key accepts UDP, TCP, and TLS.

  • custom_options: This field is reserved for setting custom options. It is typically left empty.

  • role: This field is used to specify the user’s role. It accepts the following values: StandardUser, StandardInternationalUser, QueueManager, Admin.

  • phones: This field is used to specify auto-provisioning the IP Phone for this user. It’s a JSON object that includes the following keys:

    • mac: This is the MAC address of the phone. It should be in lowercase with a separator “:” or “-”. Please ensure this Phone MAC address is not used by other users, otherwise, the provisioning will fail.

    • filename: This is the template file of the IP phone.

    • vendor: This is the vendor of the IP phone.

    • interface: This specifies the PBX IP Address of the network interface for the IP phone, which will be treated as the outbound proxy server in the IP Phone settings. The values are the same as the interface explained above.

    • preferred_transport: This specifies the PortSIP PBX transport protocol for this IP phone. The IP Phone will prioritize using this specified transport to register with PortSIP PBX. It allows UDP, TCP, or TLS.

    • model: This is the phone model.

    • password: This specifies the phone web UI password.

    • language: This specifies the display language for the IP phone. It supports the following values: ENGLISH, CHINESE, DUTCH, FRENCH, GERMAN, GREEK, ITALIAN, JAPANESE, POLISH, RUSSIAN, SPANISH, SWEDISH, UKRAINIAN, BULGARIAN.

    • timezone: This specifies the timezone for this IP Phone.

    • transfer: This specifies the transfer mode for the transfer key of the IP phone. It can be ATTENDED, BLIND, and NEW_CALL.

    • ringtone: This is the ring tone for receiving a new call.

    • queue_ringtone: This specifies the ring tone for the queue call.

    • date_format: This is the date format for the phone.

    • time_format: This is the time format for the phone.

    • powerled: This is the power LED lights.

    • backlight: This is the backlight of the IP Phone screen.

    • screensaver: This is the screensaver of the IP Phone.

    • rps: This indicates whether to store the auto-provisioning URL into the IP Phone vendor’s RPS server or not. Set to true will be stored in the RPS and false will not.

    • https: This indicates whether to generate the auto-provisioning URL with the HTTPS scheme or not. Set to true the URL will be HTTPS and false will be HTTP.

For the v22.x:

[
  {
    "mac": "0c:38:3e:63:fe:e8",
    "rps": true,
    "https": false,
    "model": "V62 Pro",
    "codecs": [
      "PCMU",
      "PCMA",
      "G729",
      "G722"
    ],
    "vendor": "Fanvil",
    "filename": "fanvil.ph.xml",
    "language": "English",
    "password": "63FEE8",
    "powerled": "Both (Voicemail and Missed calls)",
    "ringtone": "Default",
    "timezone": "GMT+8 China(Beijing)",
    "transfer": "BLIND",
    "backlight": "30 seconds",
    "interface": "WEB_DOMAIN",
    "pc_port_id": 254,
    "date_format": "15 JAN MON",
    "enable_lldp": false,
    "screensaver": "",
    "time_format": "24-hour clock",
    "wan_port_id": 256,
    "serial_number": "",
    "door_password1": "",
    "door_password2": "",
    "queue_ringtone": "Ring 1",
    "pc_port_priority": 0,
    "wan_port_priority": 0,
    "enable_vlan_pc_port": false,
    "preferred_transport": "UDP",
    "enable_vlan_wan_port": false
  }
]

For the v16.x:

[{
	"mac": "cc:5e:f8:41:b7:95",
	"filename": "yealinkT3x.ph.xml",
	"vendor": "Yealink",
	"interface": "PRIVATE_IPV4",
	"preferred_transport": "UDP",
	"model": "SIP-T32G",
	"password": "365894258",
	"language": "English",
	"timezone": "GMT-5:00 US Eastern Time, New York",
	"transfer": "BLIND",
	"ringtone": "",
	"queue_ringtone": "Ring 1",
	"date_format": "",
	"time_format": "",
	"powerled": "",
	"backlight": "",
	"screensaver": "",
	"rps": false,
	"https": false,
	"codecs": ["PCMU", "PCMA", "G729", "G722"],
	"enable_lldp": false,
	"enable_vlan_wan_port": false,
	"wan_port_id": 1,
	"wan_port_priority": 0,
	"enable_vlan_pc_port": false,
	"pc_port_id": 1,
	"pc_port_priority": 0
}]

Sample CSV File

We provide a sample CSV file for bulk importing and auto-provisioning four users.

  • For v22.x: Sample CSV file

  • For v16.x: Sample CSV file

After you download that sample CSV file, you can sign in to the PortSIP PBX web port, in the menu Call Manager > Users, click the Import button, and select the sample CSV file to import the users, the users will be successfully created with provisioning information.

Please also reference the article Zero Touch Provisioning Phones.